A how to for a PC XO.

[Llafriel]

Not sure if this has already been answered, but is it possible to use two Prodigy LT's for a total of 16 outputs?

 

[wigginjs]

As far as I can tell Console only supports one device at a time, which really sucks.

 

[JinMTVT]

JimMTVT:

Take a look at the screenshot from Vil's setup at the bottom of this post. The basic idea is to route WDM output to ASIO input in DirectWire and then turn off WDM output. Then, in Console, click the Audio button under "setup" on the main screen and select your sound card, choose ASIO, and set your sample rate to 48KHz. If you want to be able to process audio from your line-input, in Directwire connect the inputs you want to the outputs you'd like (probably 1->1, 2->2, ...). Now, in Console, click on the "Func" button to bring up the function tab. Drag out the two functions labeled "Wave In" and "Wave Out". Then connect them together with the VST plug-ins you'd like to use in between them (refer to picture below). Turn on Console using the power button and hopefully everything will work.Hope this helps.

thanks for the info
that's what i needed to know

but how do you turn off WDM output ??
i can wire easy in DWires ..

then what to do with asio output??
what is the output at the end?

 

[wigginjs]

Turning off WDM output is done in the Directwire interface. You click on the button directly under the WDM heading so that it says "off" for the output side (refer to Vil's screenshot, he has WDM output turned off). You don't need to route the output from ASIO anywhere in Directwire, again, refer to Vil's screenshot, he has Directwire configured correctly.

 

[wigginjs]

Heh, woops, no wonder you were confused. I never included a link to that screenshot I keep refering to =).

http://www.diyaudio.com/forums/attachment.php?s=&postid=650915&stamp=1117181057

 

[Llafriel]

As far as I can tell Console only supports one device at a time, which really sucks.

I'm able to use my two soundcards at the same time. one as input and the other as output. This is a onboard soundcard (Soundstorm) and my Revolution. It might be a trivial task for the developers to add support for two or more soundcards? Or perhaps as an plugin?

 

[ewildgoose]

Not sure if this has already been answered, but is it possible to use two Prodigy LT's for a total of 16 outputs?

The problem with trying to use two soundcards is that they will have slightly different clocks and hence drift apart. Imagine it was just one tick a second. After 20 mins therefore it would be 20 x 60 = 1,200 ticks misaligned - which is about 30ms at 44,100Khz (ie ticks per sec)

Pro audio cards will have a cable to let you join them together to keep them in sync (RME for example). Also another trick can be to use an spdif between the two cards which should keep them roughly aligned.

It's not trivial in other words. Have a look at the Alsa list and docs for example (linux sound drivers) there is always someone trying to do this on there. It's fairly straightforward to do under linux, but you can read pretty mixed results depending on whether they used a pro card and synced the two cards or not

It would in theory be possible to look at callback times of the two cards and resample different streams so that they go at the same rate, but this is extremely hard stuff to do correctly, and pro audio cards usually only have two soundcard buffers so you have very little leeway to get this correct.

...ok, that was the boring answer

 

[Llafriel]

I see the problem with the clocks drifting on each card. Would it be possible to sync the clock's diy style? Hardwiring one onto the other or pherhaps using an external clock?
By running two instances of console I'm able to use both my soundcards seperately, so running two Prodigys should be possible also. This of course requires that one is able to pick specific soundstreams from windows.

 

[ewildgoose]

Well of course it is. It's just an engineering problem. Doesn't mean that your software will support the two cards though. You CAN in Alsa, but I don't know about windows. Pro audio cards usually already have an external sync option, eg RME you either get the wordclock module or simply connect the two spdif ports

By the way, the front page of the wiki ( http://www.duffroomcorrection.com ) reports that there is a multichannel convolver for DirectX in development. You might want to contact the author and learn more. This approach would mean that you can calculate all your coefficients, then implement them in the DirectX sound layer and hence have access to your correction through all windows apps..

I have no idea if that's already possible using your current method of course...

 

[bser]

hey guys I"m looking to do an all active 5.1 setup for my HT project, can someone recommend the processor power I would need for 11 channels? Each speaker will be a 2 way. Also can anyone say whether the program runs better on an amd platform with the short pipes or will intel's clockspeed advantage lend any performance benefits? I'm thinking dual core amd 3800, will this processor be able to handle it? Thanks for any suggestions.

 

[ewildgoose]

Your choice of convolver is the most important question. Second is probably whether you will be resampling.

For a simple crossover, Brutefir on linux could probably do is in 400Mhz or so depending on filter lengths.

 

[ShinOBIWAN]

Goose:

I think it would be a great ideal if you could document your method for others, afterall there's many ways to skin a cat.

I think it would be a better option if cost is a first concern and you also want a high quality XO.

BTW Just to be clear:

Will this process any incoming digital or analogue audio signal?

 

[Vil]

Shin ,

if you use Foobar as audio player there is cool plugin for upsampling into 96khz . then if your computer fast enough use Console procesing at 96K . after some experiments I found it sounds better compared to 44k procesing .

Vil

 

[wigginjs]

ed:

Can't find your e-mail on the wiki and it's unavailable on this forum... How can I contact you?

 

[ewildgoose]

I think it would be a great ideal if you could document your method for others, afterall there's many ways to skin a cat.

I think it would be a better option if cost is a first concern and you also want a high quality XO.

As I said I am seriously short of time at the moment and trying to start a small business (and raise an 8 month old baby). I can give rough notes on process, and occasionally in response to a question I can scribble a more detailed reply on the wiki than I can here. But unfortunately I don't really have a bunch of time to go much beyond that.

I really need help documenting how to do this stuff on the wiki. However, you are not on your own. Sign up at the DRC page on sourceforge and there is a mailing list there where you can talk to me, Denis Sbragion (the DRC author) and a bunch of other enthusiasts. So it's not just me basically.

However, I will help you out if you have a play and get stuck. It's easier for me to answer specific questions than to write a book.

Basically though the process is.

1) Generate crossovers
2) Setup Brutefir so it takes an input signal and generates lots of outputs signals (as a result of the crossovers)
3) Then run a sweep through each channel one by one and record the output. From this generate the room IR (I use my rec_imp program to automate this - runs under windows/linux/mac)
4) Feed the room IR for each channel into DRC which generates a high quality "inversion" for that channel.
5) You then apply the correction filters to each channel (ie before the crossover)

BTW Just to be clear:

Will this process any incoming digital or analogue audio signal?

Brutefir is a pluggable system and can filter input FROM anywhere, TO anywhere. So you can filter FROM a file, TO the soundcard. Or FROM the soundcard TO a file.

In my case I use the Jack audio layer under windows so that the output of any of my audio applications gets fed directly into Brutefir. The OUTPUT of brutefir is always the soundcard (and hence the speakers).

There is no reason why I could not also *simultaneously* be filtering the input of my soundcard and mixing it into the output. That way I can have (simultaneously if required) the audio generated from the HTPC, and also any external audio on the digital or analogue inputs filtered and processed to the card output

So "yes". You can do just about anything

The only thing which doesn't look like it has been written (and I might knock something up) is a way to take dolby or DTS surround IN on the spdif, decode it, and then pass it onto Brutefir for filtering. This shouldn't be hard to write, but it's not done as far as I know

Good luck

P.S. Email is something like drcstuff (at) wildgooses (dot) com

 

[Audiophilenoob]

shin would another alternative to the clock you use be something like this:

http://www.poweralley.com/quantum_ii.htm

I'm more curious if this will work for a 3-way 2 channel stereo and xovers rather than being necessarily cheaper... I'm just wondering if this will work so I can look for other options

 

[ewildgoose]

That just looks like a cheap EQ box?

Your problem is that after a while one card is going to be sending out an analogue signal which might in theory be even seconds ahead of the other card. Actually this will never happen because all the buffering will fall apart once the cards skew more than a few hundred millisecs, but the idea is there

You need to physically sync the soundcards themselves. Some cards you can do this on the cheap with just an spdif between them. Others need some external clock signal. It's not rocket science, you just have to do it

Even easier though is probably just to buy a DAC with lots of channels...

If your problem is getting enough outputs for crossover duties then why not look at a Behringer DCX2496 or similar. It's cheap and cheerful and gets you started for practically nothing. Its easy to setup and basically handles all the crossover stuff, at least as long as you don't mind simple emulations of the normal analogue crossovers like LR, etc

This leaves you only having to do DRC on the PC and also a much simpler system with less chance of blowing a tweeter due to a software cockup...

 

[Audiophilenoob]

That just looks like a cheap EQ box?

it's a processor... read about it

it's also $2199 so no not cheap either... and it can be used with wordclock

 

[JinMTVT]

this is starting to annoyed me seriosuly ..

i've got all the softwares shinob talked about
( testing for now..if i like i'll get what i need! )

My soundcard is an ESI-PRO Waveterminal 192L
wich got e-WDM drivers ( asio, WDM, MME,GSIF whatever..)
+directwires built in the drivers

so i tried to setup Console to get some sound IN
at least ..showing on the CurvesEQ graph

i tried every possible combination of audio settings in console
and directwires stuff ..

i tried extensivly to get anything using ASIO in console
and in directwires setup as seen on previous pics
without anything showing up at all in curvesEQ

then, if i set up in console for direct sound i get something ine CurvesEQ wich is my line input ( MIC input )
any other input doesn't do a thing at all

is there something i ain't getting right here?

i tried with mediaplayer playing a cd audio file
then foobar with almost all possible output
( except for asio, wich doesn't seem to load at all in foobar)
may my asio is broken ? how to check if it is working?

i don't konw much about audio specific aps in PC
( asio and stuff. )

please help me out, i just wanna set this thing up and try the DRC
to see how i like it!!

thanks

 

[wigginjs]

Post a pic showing your console setup and directwire, etc. Also show the console audio setup tab in the pic. I'll try to figure out whats going on from there.